Field of the Disclosure
The technology of the disclosure relates generally to Web Real-Time Communications (WebRTC) interactive sessions utilizing Session Initiation Protocol (SIP) signaling.
Technical Background
Web Real-Time Communications (WebRTC) is an ongoing effort to develop industry standards for integrating real-time communications functionality into web clients, such as web browsers, to enable direct interaction with other web clients. This real-time communications functionality is accessible by web developers via standard markup tags, such as those provided by version 5 of the Hyper Text Markup Language (HTML5), and client-side scripting Application Programming Interfaces (APIs) such as JavaScript APIs. More information regarding WebRTC may be found in “WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web,” by Alan B. Johnston and Daniel C. Burnett, 2nd Edition (2013 Digital Codex LLC), which is incorporated in its entirety herein by reference.
WebRTC provides built-in capabilities for establishing real-time video, audio, and/or data streams in both point-to-point interactive sessions and multi-party interactive sessions. The WebRTC standards are currently under joint development by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). Information on the current state of WebRTC standards can be found at, e.g., http://www.w3c.org and http://www.ietf.org.
In a typical WebRTC exchange, two WebRTC clients retrieve WebRTC web applications, such as HTML5/JavaScript web applications, from a web application server. Through the web applications, the two WebRTC clients engage in an initiation dialogue for initiating a peer connection over which a WebRTC interactive flow (e.g., a real-time video, audio, and/or data exchange) will pass. This initiation dialogue may include a media negotiation used to communicate and reach an agreement on parameters that define characteristics of the WebRTC interactive flow. In some embodiments, the media negotiation may be implemented via a WebRTC offer/answer exchange (using, e.g., Session Description Protocol (SDP) objects) via a secure network connection such as a Hyper Text Transfer Protocol Secure (HTTPS) connection or a Secure WebSockets connection. Once the initiation dialogue is complete, the WebRTC clients may then establish a direct peer connection with one another, and may begin an exchange of media or data packets transporting real-time communications.
While WebRTC provides the media capabilities necessary for real-time communications, it does not specify a call signaling mechanism to be used for a WebRTC interactive flow. Accordingly, the responsibility for implementing call signaling for the WebRTC interactive flow falls to the WebRTC web application. One approach for call signaling is to employ a Session Initiation Protocol (SIP) user agent, implemented either at a WebRTC endpoint or at an intermediate web server. However, a SIP user agent implemented at a WebRTC endpoint may require the use of client-side scripting code that may be viewed and/or manipulated by an end user. As a result, intellectual property within the SIP user agent may be compromised, or the code constituting the SIP user agent may be manipulated for malicious purposes. A SIP user agent implemented at an intermediate web server may avoid these issues, but may introduce new challenges. For instance, the SIP user agent at the web server may need to constantly maintain a state of an ongoing WebRTC communication. This may violate the stateless nature of the web server, and may result in load-balancing and/or reliability problems for the web server.